[FFmpeg-soc] [PATCH] AMR-WB Decoder
Marcelo Galvão Póvoa
marspeoplester at gmail.com
Fri Aug 27 23:13:57 CEST 2010
On 26 August 2010 18:55, Vitor Sessak <vitor1001 at gmail.com> wrote:
> Marcelo Galvão Póvoa wrote:
>>
>> Fixed excitation array incorrect length.
>>
>> Should now be applicable cleanly to the latest ffmpeg revision
>> (fd151a5f8bd152c456a).
>
> First look:
>
>> diff --git a/libavcodec/Makefile b/libavcodec/Makefile
>> index 1422b5c..9c68d72 100644
>> --- a/libavcodec/Makefile
>> +++ b/libavcodec/Makefile
>> @@ -60,6 +60,7 @@ OBJS-$(CONFIG_AMRNB_DECODER) += amrnbdec.o
>> celp_filters.o \
>> celp_math.o acelp_filters.o \
>> acelp_vectors.o \
>> acelp_pitch_delay.o
>> +OBJS-$(CONFIG_AMRWB_DECODER) += amrwbdec.o
>
> You are using a lot of functions (ff_acelp_interpolatef, ff_lsp2polyf,
> etc) that are not in amrwbdec.c, so you need to add the dependencies of
> everything you use. To see if you didn't forgot any try "./configure
> --disable-decoders --enable-decoder=amr-wb".
>
Fixed and tested
>> +static av_cold int amrwb_decode_init(AVCodecContext *avctx)
>> +{
>> + AMRWBContext *ctx = avctx->priv_data;
>> + int i;
>> +
>> + avctx->sample_fmt = SAMPLE_FMT_FLT;
>> +
>> + av_lfg_init(&ctx->prng, 1);
>> +
>> + ctx->excitation = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER
>> + 1];
>> + ctx->first_frame = 1;
>
>> + ctx->tilt_coef = ctx->prev_tr_gain = 0.0;
>
> not needed, ctx is already allocated with av_mallocz().
>
Fixed
>> +/**
>> + * Parses a speech frame, storing data in the Context
>> + *
>> + * @param[in,out] ctx The context
>> + * @param[in] buf Pointer to the input buffer
>> + * @param[in] buf_size Size of the input buffer
>> + *
>> + * @return The frame mode
>> + */
>> +static enum Mode unpack_bitstream(AMRWBContext *ctx, const uint8_t *buf,
>> + int buf_size)
>> +{
>> + GetBitContext gb;
>> + enum Mode mode;
>> + uint16_t *data;
>> +
>> + init_get_bits(&gb, buf, buf_size * 8);
>> +
>> + /* decode frame header (1st octet) */
>> + skip_bits(&gb, 1); // padding bit
>> + ctx->fr_cur_mode = get_bits(&gb, 4);
>
>> + mode = ctx->fr_cur_mode;
>
> Why the extra "mode" variable? You can just use ctx->fr_cur_mode directly...
>
It was used to improve readability but looking now I guess it may be
discarded. Fixed
>> + if (mode < MODE_SID) { /* Normal speech frame */
>> + const uint16_t *perm = amr_bit_orderings_by_mode[mode];
>> + int field_size;
>> +
>> + while ((field_size = *perm++)) {
>> + int field = 0;
>> + int field_offset = *perm++;
>> + while (field_size--) {
>> + uint16_t bit_idx = *perm++;
>> + field <<= 1;
>> + /* The bit index inside the byte is reversed (MSB->LSB) */
>> + field |= BIT_POS(buf[bit_idx >> 3], 7 - (bit_idx & 7));
>> + }
>> + data[field_offset] = field;
>> + }
>> + }
>> + else if (mode == MODE_SID) { /* Comfort noise frame */
>> + /* not implemented */
>> + }
>
> av_log_missing_feature(...);
>
It is already called at the main decode routine. I will just remove
this else statement.
>> +/**
>> + * Convert an ISF vector into an ISP vector
>> + *
>> + * @param[in] isf Isf vector
>> + * @param[out] isp Output isp vector
>> + * @param[in] size Isf/isp size
>> + */
>> +static void isf2isp(const float *isf, double *isp, int size)
>> +{
>> + int i;
>> +
>> + for (i = 0; i < size - 1; i++)
>> + isp[i] = cos(2.0 * M_PI * isf[i]);
>> +
>> + isp[size - 1] = cos(4.0 * M_PI * isf[size - 1]);
>> +}
>
> Almost duplication of amrnbdec.c:lsf2lsp().
>
Yes, but this last element doubling diverges between AMR-NB and AMR-WB
reference sources.
>> +/**
>> + * Decodes quantized ISF vectors using 36-bit indices (6K60 mode only)
>> + *
>> + * @param[in] ind Array of 5 indices
>> + * @param[out] isf_q Buffer for isf_q[LP_ORDER]
>> + * @param[in] fr_q Frame quality (good frame == 1)
>> + *
>> + */
>> +static void decode_isf_indices_36b(uint16_t *ind, float *isf_q, uint8_t
>> fr_q) {
>> + int i;
>> +
>> + if (fr_q == 1) {
>> + for (i = 0; i < 9; i++) {
>> + isf_q[i] = dico1_isf[ind[0]][i] / (float) (1 << 15);
>
> isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
>
> It is faster and the compiler cannot do it for you (this applies to a
> lot of code in your patch).
>
Fixed everywhere
>> + for (i = 0; i < 7; i++) {
>> + isf_q[i + 9] = dico2_isf[ind[1]][i] / (float) (1 << 15);
>> + }
>> + for (i = 0; i < 5; i++) {
>> + isf_q[i] += dico21_isf_36b[ind[2]][i] / (float) (1 << 15);
>> + }
>> + for (i = 0; i < 4; i++) {
>> + isf_q[i + 5] += dico22_isf_36b[ind[3]][i] / (float) (1 <<
>> 15);
>> + }
>> + for (i = 0; i < 7; i++) {
>> + isf_q[i + 9] += dico23_isf_36b[ind[4]][i] / (float) (1 <<
>> 15);
>> + }
>> + }
>> + /* not implemented for bad frame */
>
> Unless you plan to implement bad frame handling soon, I suggest you just
> remove the function parameter. It serves to nothing as now.
>
Fixed
>> +/**
>> + * Ensures a minimum distance between adjacent ISFs
>> + *
>> + * @param[in,out] isf ISF vector
>> + * @param[in] min_spacing Minimum gap to keep
>> + * @param[in] size ISF vector size
>> + *
>> + */
>> +static void isf_set_min_dist(float *isf, float min_spacing, int size) {
>> + int i;
>> + float prev = 0.0;
>> +
>> + for (i = 0; i < size - 1; i++) {
>> + isf[i] = FFMAX(isf[i], prev + min_spacing);
>> + prev = isf[i];
>> + }
>> +}
>
> Please do not duplicate lsp.c:ff_set_min_dist_lsf().
>
Oops, I missed that. Fixed
>> +/**
>> + * 16kHz version of ff_lsp2polyf for the high-band
>> + */
>> +static void lsp2polyf_16k(const double *lsp, double *f, int
>> lp_half_order)
>> +{
>> + int i, j;
>> +
>> + f[0] = 0.25;
>> + f[1] = -0.5 * lsp[0];
>> + lsp -= 2;
>> + for(i = 2; i <= lp_half_order; i++)
>> + {
>> +
>
> nit:
> for(i = 2; i <= lp_half_order; i++) {
>
Fixed
>> + f[i] = val * f[i - 1] + 2 * f[i - 2];
>> + for(j = i - 1; j > 1; j--)
>> + f[j] += f[j - 1] * val + f[j - 2];
>> + f[1] += 0.25 * val;
>> + }
>> +}
>
> Hmm, can't this function be replaced by ff_lsp2polyf() with some rescaling?
>
Actually I realize now that the difference between these functions is
just the entire output scaled down by 0.25. That was silly because I
was scaling by 4.0 after this function. It is weird why the reference
decoder does this process. Maybe it should worth someone checking it
out to be sure about this.
>> +/**
>> + * Convert a ISP vector to LP coefficient domain {a_k}
>> + * Equations from TS 26.190 section 5.2.4
>> + *
>> + * @param[in] isp ISP vector for a subframe
>> + * @param[out] lp LP coefficients
>> + * @param[in] lp_half_order Half the number of LPs to construct
>> + */
>> +static void isp2lp(const double *isp, float *lp, int lp_half_order) {
>> + double pa[10 + 1], qa[10 + 1];
>> + double last_isp = isp[2 * lp_half_order - 1];
>> + double qa_old = 0.0;
>> + float *lp2 = &lp[2 * lp_half_order];
>> + int i;
>> +
>> + if (lp_half_order > 8) { // high-band specific
>> + lsp2polyf_16k(isp, pa, lp_half_order);
>> + lsp2polyf_16k(isp + 1, qa, lp_half_order - 1);
>> +
>> + for (i = 0; i <= lp_half_order; i++)
>> + pa[i] *= 4.0;
>> + for (i = 0; i < lp_half_order; i++)
>> + qa[i] *= 4.0;
>> + } else {
>> + ff_lsp2polyf(isp, pa, lp_half_order);
>> + ff_lsp2polyf(isp + 1, qa, lp_half_order - 1);
>> + }
>> +
>> + for (i = 1; i < lp_half_order; i++) {
>> + double paf = (1 + last_isp) * pa[i];
>> + double qaf = (1 - last_isp) * (qa[i] - qa_old);
>> +
>> + qa_old = qa[i - 1];
>> +
>> + lp[i] = 0.5 * (paf + qaf);
>> + lp2[-i] = 0.5 * (paf - qaf);
>> + }
>> +
>> + lp[0] = 1.0;
>> + lp[lp_half_order] = 0.5 * (1 + last_isp) * pa[lp_half_order];
>> + lp2[0] = last_isp;
>> +}
>
> Please double-check if this is not a duplication of sipr.c:lsp2lpc_sipr().
>
Interesting, I believe they are the same function except for the fact
that in AMR-WB the first LP coefficient is always 1.0 but this should
be easily adapted by passing lp + 1 to the function and setting lp[0]
= 1.0. Also, the LP order is different between these codecs. To reuse
this existing function it should be moved to another file (lsp.c I
guess) and modified. Should I do it and include in the next patch?
>
>> +static void decode_pitch_vector(AMRWBContext *ctx,
>> + const AMRWBSubFrame *amr_subframe,
>> + const int subframe)
>> +{
>> + int pitch_lag_int, pitch_lag_frac;
>> + int i;
>> + float *exc = ctx->excitation;
>> + enum Mode mode = ctx->fr_cur_mode;
>> +
>> + if (mode <= MODE_8k85) {
>> + decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac,
>> amr_subframe->ada
>> p,
>> + &ctx->base_pitch_lag, subframe, mode);
>> + } else
>> + decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac,
>> amr_subframe->ad
>> ap,
>> + &ctx->base_pitch_lag, subframe);
>> +
>> + ctx->pitch_lag_int = pitch_lag_int;
>> + pitch_lag_int += (pitch_lag_frac < 0 ? -1 : 0) + (pitch_lag_frac
>> ? 1 : 0);
>> +
>> +
>> + /* Calculate the pitch vector by interpolating the past excitation at
>> the
>> + pitch lag using a hamming windowed sinc function */
>> + ff_acelp_interpolatef(exc, exc + 1 - pitch_lag_int,
>> + ac_inter, 4,
>> + pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),
>> + LP_ORDER, AMRWB_SFR_SIZE + 1);
>
> ac_inter is yet another hamming function. Can you check if you cannot reuse
> acelp_vectors.c:ff_b60_sinc or sipr16kdata.h:sinc_win?
>
Well, I cannot tell precisely if it cannot be reused but I guess so.
The number of coefficients in the table is related to the fractional
resolution of the interpolation: in AMR-NB and SIPR it is 4 but in
AMR-WB it is 6. Also, I don't see similarities between the table I
extracted from the reference and these tables.
>> +/**
>> + * Reduce fixed vector sparseness by smoothing with one of three IR
>> filters
>> + * Also known as "adaptive phase dispersion"
>> + *
>> + * @param[in] ctx The context
>> + * @param[in,out] fixed_vector Unfiltered fixed vector
>> + * @param[out] buf Space for modified vector if necessary
>> + *
>> + * @return The potentially overwritten filtered fixed vector address
>> + */
>> +static float *anti_sparseness(AMRWBContext *ctx,
>> + float *fixed_vector, float *buf)
>
> amrnbedec.c has a function with the same name. Any code can be reused?
>
The function skeletons are similar but there are differences
everywhere. I can think about easily reusing only about 6 lines, which
wouldn't worth much
>> +/**
>> + * Apply to synthesis a 2nd order high-pass filter
>> + *
>> + * @param[out] out Buffer for filtered output
>> + * @param[in] hpf_coef Filter coefficients as used below
>> + * @param[in,out] mem State from last filtering (updated)
>> + * @param[in] in Input speech data
>> + *
>> + * @remark It is safe to pass the same array in in and out parameters
>> + */
>> +static void high_pass_filter(float *out, const float hpf_coef[2][3],
>> + float mem[4], const float *in)
>> +{
>> + int i;
>> + float *x = mem - 1, *y = mem + 2; // previous inputs and outputs
>> +
>> + for (i = 0; i < AMRWB_SFR_SIZE; i++) {
>> + float x0 = in[i];
>> +
>> + out[i] = hpf_coef[0][0] * x0 + hpf_coef[1][0] * y[0] +
>> + hpf_coef[0][1] * x[1] + hpf_coef[1][1] * y[1] +
>> + hpf_coef[0][2] * x[2];
>> +
>> + y[1] = y[0];
>> + y[0] = out[i];
>> +
>> + x[2] = x[1];
>> + x[1] = x0;
>> + }
>> +}
>
> acelp_filter.c:ff_acelp_apply_order_2_transfer_function()
>
Are you sure? I can't see them as equivalent, could you explain?
>> +/**
>> + * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using
>> + * a FIR interpolation filter. Uses past data from before *in address
>> + *
>> + * @param[out] out Buffer for interpolated signal
>> + * @param[in] in Current signal data (length
>> 0.8*o_size)
>> + * @param[in] o_size Output signal length
>> + */
>> +static void upsample_5_4(float *out, const float *in, int o_size)
>> +{
>> + const float *in0 = in - UPS_FIR_SIZE + 1;
>> + int i;
>> +
>> + for (i = 0; i < o_size; i++) {
>> + int int_part = (4 * i) / 5;
>> + int frac_part = (4 * i) - 5 * int_part;
>
> You can break this loop in two to avoid the division:
>
> i = 0;
> for (j = 0; j < o_size/5; j++)
> for (k = 0; k < 5; k++) {
> ....
> i++;
> }
>
I've done this in not so neat way, adding 4 and calculating a simple
remainder by 5. Tell me if can be done in a better way.
>
>> +/**
>> + * Generate the high-band excitation with the same energy from the lower
>> + * one and scaled by the given gain
>> + *
>> + * @param[in] ctx The context
>> + * @param[out] hb_exc Buffer for the excitation
>> + * @param[in] synth_exc Low-band excitation used for synthesis
>> + * @param[in] hb_gain Wanted excitation gain
>> + */
>> +static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
>> + const float *synth_exc, float hb_gain)
>> +{
>> + int i;
>> + float energy = ff_dot_productf(synth_exc, synth_exc, AMRWB_SFR_SIZE);
>> +
>> + /* Generate a white-noise excitation */
>> + for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
>> + hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);
>> +
>> + ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc, energy,
>> + AMRWB_SFR_SIZE_16k);
>> +
>> + for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
>> + hb_exc[i] *= hb_gain;
>> +}
>
> Why are you scaling it twice?
>
The first scaling is to match the energy with the lower band part. The
second is the high_band gain itself. This can be done in only one loop
using ff_scale_vector_to_given_sum_of_squares() ?
>> +/**
>> + * Apply to high-band samples a 15th order filter
>> + * The filter characteristic depends on the given coefficients
>> + *
>> + * @param[out] out Buffer for filtered output
>> + * @param[in] fir_coef Filter coefficients
>> + * @param[in,out] mem State from last filtering (updated)
>> + * @param[in] cp_gain Compensation gain (usually the filter
>> gain)
>> + * @param[in] in Input speech data (high-band)
>> + *
>> + * @remark It is safe to pass the same array in in and out parameters
>> + */
>> +static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE +
>> 1],
>> + float mem[HB_FIR_SIZE], float cp_gain, const
>> float *in)
>> +{
>> + int i, j;
>> + float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current
>> samples
>> +
>> + memcpy(data, mem, HB_FIR_SIZE * sizeof(float));
>> +
>> + for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
>> + data[i + HB_FIR_SIZE] = in[i] / cp_gain;
>> +
>> + for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) {
>> + out[i] = 0.0;
>> + for (j = 0; j <= HB_FIR_SIZE; j++)
>> + out[i] += data[i + j] * fir_coef[j];
>> + }
>> +
>> + memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float));
>> +}
>
> I think it is cleaner (and more consistent) to do like the synthesis filter
> and get one single buffer for output and memory...
>
The problem is that this filter memory consists of past inputs and not
outputs. If I would do with a single buffer I would have to copy the
input to it in order to avoid overwriting the last filter output.
--
Marcelo
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